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Gutschy
16.03.06, 14:16
Moin Leute,

poste hier aus reiner Verzweifelung, vielleicht wissen ja die Netzwerk Cracks Rat.

kphone funktionierte erst 2 Tage. Die Einstellung hab ich mir hier abgeschrieben.
http://www.ip-phone-forum.de/showthread.php?t=70600
Und jetzt, ohne einen Grund kommt meine Stimme an der Gegenstelle nur noch zerhackt an. Einstellungen wurden nicht verändert. Das gibt Kphone über Konsole folgendes aus.


Found 2 interfaces.
SipClient: Listening UDP on port: 5060
SipClient: Our address: 192.168.0.3
SipClient: STUN request
SipClient: Receiving message...
SipClient: STUN response
address_port: 33638
address: 80.143.74.25
KCallWidget: Switching calls...
CallAudio: listening for incomming RTP
UDPMessageSocket: Listening on 32920
DspOutRtp: STUN request
SipClient: Empfange Nachricht ...
SipClient: STUN response for RTP
CallAudio: Opening ALSA device for Output



----------<170 - 2730>--------------


CallAudio: Creating RTP->ALSA Diverter

SipClient: Sending: 12:44:40.414
--------------------------------
INVITE sip:02572958008@sip.web.de SIP/2.0
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4D213DD8
CSeq: 2900 INVITE
To: <sip:02572958008@sip.web.de>
Content-Type: application/sdp
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
Call-ID: 1976153876@80.143.74.25
Subject: sip:mcgutschy@sip.web.de
Content-Length: 228
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:mcgutschy@80.143.74.25:33638;transport=udp>

v=0
o=username 0 0 IN IP4 80.143.74.25
s=The Funky Flow
c=IN IP4 80.143.74.25
t=0 0
m=audio 33127 RTP/AVP 0 97 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

SipClient: Sending to 'sip.web.de:5060'
SipClient: Receiving message...

SipClient: Received: 12:44:40.583
---------------------------------
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4D213DD8;rport=33 638
CSeq: 2900 INVITE
To: <sip:02572958008@sip.web.de>;tag=f43d8ce4e4130f38b 15965d9884f209e.9634
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
Call-ID: 1976153876@80.143.74.25
Proxy-Authenticate: Digest realm="web.de", nonce="4416ad5f58fb47d043861ab78b7ef9
f420df4925"
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 sip-ha.web.de:5060 "Noisy feedback tells: pid=23142 req_src_ip=80.
143.74.25 req_src_port=33638 in_uri=sip:02572958008@sip.web.de out_uri=sip:02572
958008@sip.web.de via_cnt==1"


SipCall: Incoming response
SipTransaction: Incoming Response

SipClient: Sending: 12:44:40.583
--------------------------------
ACK sip:02572958008@sip.web.de SIP/2.0
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4D213DD8
CSeq: 2900 ACK
To: <sip:02572958008@sip.web.de>;tag=f43d8ce4e4130f38b 15965d9884f209e.9634
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
Call-ID: 1976153876@80.143.74.25
Content-Length: 0
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:mcgutschy@80.143.74.25:33638;transport=udp>


SipClient: Sending to 'sip.web.de:5060'
SipCallMember: localStatusUpdated: 407
WL: SipProtocol: HA1=1c9f70a6aa2b8ccb6719ba8628d2a976 (mcgutschy:web.de)
SipProtocol: Digest calculated.

SipClient: Sending: 12:44:40.646
--------------------------------
INVITE sip:02572958008@sip.web.de SIP/2.0
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4328A13B
CSeq: 2901 INVITE
To: <sip:02572958008@sip.web.de>
Proxy-Authorization: Digest username="mcgutschy", realm="web.de", nonce="4416ad5
f58fb47d043861ab78b7ef9f420df4925", uri="sip:02572958008@sip.web.de", cnonce="ab
cdefghi", nc=00000001, response="38dcf9f7fa28cf26d1376a2b1412ee4f", opaque="", a
lgorithm="MD5"
Content-Type: application/sdp
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
Call-ID: 1976153876@80.143.74.25
Subject: sip:mcgutschy@sip.web.de
Content-Length: 228
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:mcgutschy@80.143.74.25:33638;transport=udp>

v=0
o=username 0 0 IN IP4 80.143.74.25
s=The Funky Flow
c=IN IP4 80.143.74.25
t=0 0
m=audio 33127 RTP/AVP 0 97 8 3
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30

SipClient: Sending to 'sip.web.de:5060'
SipClient: Receiving message...

SipClient: Received: 12:44:40.859
---------------------------------
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4328A13B;rport=33 638
CSeq: 2901 INVITE
To: <sip:02572958008@sip.web.de>
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
Call-ID: 1976153876@80.143.74.25
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 sip-ha.web.de:5060 "Noisy feedback tells: pid=23135 req_src_ip=80.
143.74.25 req_src_port=33638 in_uri=sip:02572958008@sip.web.de out_uri=sip:02572
958008@sip.web.de via_cnt==1"


SipCall: Incoming response
SipTransaction: Incoming Response
SipCallMember: localStatusUpdated: 100
SipClient: Receiving message...

SipClient: Received: 12:44:40.927
---------------------------------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 80.143.74.25:33638;rport=33638;branch=z9hG4bK4328A 13B
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
To: <sip:02572958008@sip.web.de>;tag=as30019196
Call-ID: 1976153876@80.143.74.25
CSeq: 2901 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:02572958008@217.72.200.72:5060>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 4820 4820 IN IP4 217.72.200.72
s=session
c=IN IP4 217.72.200.72
t=0 0
m=audio 10636 RTP/AVP 8 0 111 3 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=silenceSuppff - - - -

SipCall: Incoming response
SipTransaction: Incoming Response
SipCallMember: localStatusUpdated: 183
CallAudio: Using G711a for output
CallAudio: Sende an die Gegenstelle 217.72.200.72:10636
CallAudio: Opening ALSA device for Input



----------<170 - 2730>--------------


CallAudio: Creating ALSA->RTP Diverter
SipClient: Receiving message...

SipClient: Received: 12:44:42.967
---------------------------------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 80.143.74.25:33638;rport=33638;branch=z9hG4bK4328A 13B
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
To: <sip:02572958008@sip.web.de>;tag=as30019196
Call-ID: 1976153876@80.143.74.25
CSeq: 2901 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:02572958008@217.72.200.72:5060>
Content-Length: 0


SipCall: Incoming response
SipTransaction: Incoming Response
SipCallMember: localStatusUpdated: 180
CallAudio: Using G711a for output
SipClient: Receiving message...

SipClient: Received: 12:44:47.391
---------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.143.74.25:33638;rport=33638;branch=z9hG4bK4328A 13B
Record-Route: <sip:217.72.200.89;ftag=6154D669;lr=on>
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
To: <sip:02572958008@sip.web.de>;tag=as30019196
Call-ID: 1976153876@80.143.74.25
CSeq: 2901 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:02572958008@217.72.200.72:5060>
Content-Type: application/sdp
Content-Length: 264

v=0
o=root 4820 4821 IN IP4 217.72.200.72
s=session
c=IN IP4 217.72.200.72
t=0 0
m=audio 10636 RTP/AVP 8 0 111 3 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=silenceSuppff - - - -

SipCall: Incoming response
SipCall: Checking for Contact and Record-Route
SipCall: Setting Contact for this Call Member
SipTransaction: Incoming Response

SipClient: Sending: 12:44:47.392
--------------------------------
ACK sip:02572958008@217.72.200.72:5060 SIP/2.0
Via: SIP/2.0/UDP 80.143.74.25:33638;branch=z9hG4bK4328A13B
CSeq: 2901 ACK
To: <sip:02572958008@sip.web.de>;tag=as30019196
From: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
Call-ID: 1976153876@80.143.74.25
Route: <sip:217.72.200.89;ftag=6154D669;lr=on>
Content-Length: 0
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:mcgutschy@80.143.74.25:33638;transport=udp>


SipClient: Sending to 'sip.web.de:5060'
SipCallMember: localStatusUpdated: 200
CallAudio: Using G711a for output
dtmfsenderTimeout
SipClient: Receiving message...

SipClient: Received: 12:45:10.273
---------------------------------
BYE sip:mcgutschy@80.143.74.25:33638 SIP/2.0
Record-Route: <sip:217.72.200.89;ftag=as30019196;lr=on>
Via: SIP/2.0/UDP 217.72.200.89;branch=z9hG4bKd9df.b50a6c17.0
Via: SIP/2.0/UDP 217.72.200.72:5060;branch=z9hG4bK287036f3;rport=50 60
From: <sip:02572958008@sip.web.de>;tag=as30019196
To: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
Contact: <sip:02572958008@217.72.200.72:5060>
Call-ID: 1976153876@80.143.74.25
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 16
Content-Length: 0
P-hint: rr-enforced


SipCall: Incoming request
SipCall: New transaction created
SipTransaction: Incoming Request
SipClient: Sending UDP Response
SipClient: Sending to '217.72.200.89' port 5060

SipClient: Sending: 12:45:10.273
--------------------------------
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.72.200.89;branch=z9hG4bKd9df.b50a6c17.0
Via: SIP/2.0/UDP 217.72.200.72:5060;rport=5060;branch=z9hG4bK287036 f3
From: <sip:02572958008@sip.web.de>;tag=as30019196
CSeq: 102 BYE
Call-ID: 1976153876@80.143.74.25
To: "Michael Holz" <sip:mcgutschy@sip.web.de>;tag=6154D669
Content-Length: 0
User-Agent: kphone/4.2
Contact: "Michael Holz" <sip:mcgutschy@80.143.74.25:33638;transport=udp>
Record-Route: <sip:217.72.200.89;ftag=as30019196;lr=on>


KCallWidget: Starting force disconnect...
SipClient: STUN request
SipClient: Receiving message...
SipClient: STUN response
address_port: 33638
address: 80.143.74.25
[Gutschy@localhost ~]$